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279 commits

Author SHA1 Message Date
Olivier Crête
2e54d38dd0 rtpsession: Keep local conflicting addresses in the session 
As we now replace the local RTPSource on a conflict, it's no longer possible
to keep local conflicts in the RTPSource, so they instead need to be kept
in the RTPSession.

Also fix the rtpcollision test to generate multiple collisions instead of
one by change the address, as otherwise we detected that it was a single one.
 2014-05-03 18:30:20 -04:00
Wim Taymans
38a486b374 rtpsession: send reconfigure when internal-ssrc changes 
When the internal-ssrc property changes, we want to send a reconfigure
upstream to make payloaders use the new suggested ssrc.
Using the internal-ssrc property to change the SSRC of a stream is not a
good idea and doesn't work when there are multiple senders, we want to
set the SSRC directly on the payloaders. Therefore, deprecate this
property.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725361
 2014-04-18 10:21:27 +02:00
Wim Taymans
d2f93e3afc session: small cleanups 
It's nicer to explicitly check for NULL on pointer types to make it
clear that it's a pointer and not a boolean.
 2014-03-05 14:28:26 +01:00
Wim Taymans
5818a0de1a session: handle unknown SSRC in FIR 
https://bugzilla.gnome.org/show_bug.cgi?id=725712
 2014-03-05 14:27:47 +01:00
Wim Taymans
03e4a180da session: place SSRC in Retransmission event 2014-01-03 20:48:29 +01:00
Wim Taymans
ee7f41ba2e rtpsession: internal-ssrc is no longer deprecated 2013-12-30 17:00:45 +01:00
George Kiagiadakis
17517ca491 rtpsession: allow setting internal-ssrc again 2013-12-30 14:03:05 +01:00
Olivier Crête
ada6ea668b rtpsession: Add error message if the app tries to set the internal-ssrc 2013-12-13 17:36:36 -05:00
Olivier Crête
d715010d78 rtpsession: Only count nacks when a nack packet is received 
Not when any RTCP feedback packet is.
 2013-12-13 16:08:35 -05:00
Olivier Crête
7af9fdbca6 rtpsession: Process PSFB FIR requests which lack the media ssrc 
According to RFC 5104 section 4.3.1.2, RTCP PSFB FIR message SHALL
have a media_ssrc field set to 0. The actual media ssrc is in the FCI.
So in that case, we ignore the retained feedback and just let it through
to the rtp_session_process_fir() function which will check for the actual
SSRC inside the FCI.

Fixes a regression introduced by commit 57c27ec3
 2013-12-13 16:01:07 -05:00
George Kiagiadakis
6a2de911fa rtpsession: fix rb blocks disappearing after the first rtcp cycle with multiple senders 
Previously, when the session had multiple internal sender SSRCs, it would
issue SR reports with RB blocks only on the first RTCP timeout and afterwards
SR reports would be sent empty. This was because the "generation" number
in RTPSource would increase more than once during the same cycle and afterwards
it would always be greater than the session's generation, which would cause
it to be skipped from being included in RBs.

This commit fixes this problem by:
1) Increasing the RTPSource generation only at the end of each cycle,
which essentially fixes the problem but only when the internal senders
are less than GST_RTCP_MAX_RB_COUNT.
2) Keeping for each RTPSource a set of SSRCs which stores which SSRC's
SR the given RTPSource has been reported in, which also fixes the problem
when the internal senders are more than GST_RTCP_MAX_RB_COUNT. This is
necessary because of the fact that any RTPSource is marked as reported
in itself's SR and makes it impossible to know if it has been reported
in other SRs too or not, and which.
 2013-12-12 16:44:27 +01:00
George Kiagiadakis
c78a115154 rtpsession: keep extra stats for scheduling BYE 
Keep an extra stats structure for scheduling the BYE packets. When we
decide to schedule BYE, make a copy of the current stats into the
bye_stats. Then while we schedule the BYE, update and use only the
bye_stats. When we finished scheduling the BYE packet, we use the
regular stats again.
 2013-12-12 10:38:43 +01:00
George Kiagiadakis
282028e753 rtpsession: when we schedule BYE, only deal with BYE sources 
When we are doing the RTCP timeout to schedule BYE packets, don't
generate RTCP for all sources but only for the sources marked as BYE.
 2013-12-12 10:34:38 +01:00
George Kiagiadakis
6a421c3d81 rtpsession: reset state after scheduling BYE 
After we do RTCP, we are not scheduling bye anymore.
 2013-12-12 10:32:48 +01:00
George Kiagiadakis
0a0ff100ef rtpsession: also count NACKS when no signal was pending 2013-12-12 10:31:38 +01:00
George Kiagiadakis
bec9c04ea0 session: ignore RTCP packets for the BYE sources 
When we are scheduling BYE packets, ignore all RTCP for the sources that
are scheduling a BYE packet. Other sources that are not scheduling BYE
should continue receiving RTCP packets as usual.
 2013-12-12 10:09:25 +01:00
Julien Isorce
33b398e345 rtpsession: determine if the session is doing point-to-point 
In this case T_dither_max is set to 0 according to RFC 4585
 2013-12-10 16:57:56 +01:00
Wim Taymans
e8edecc56e rtpsession: don't unref buffer twice 
Cleaning the packet info will already unref the buffer.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=715078
 2013-11-28 16:51:13 +01:00
Tim-Philipp Müller
548e756e0a rtpmanager: fix Since markers 
Should be next stable release series version
 2013-11-16 12:15:14 +00:00
Torrie Fischer
acf74435e3 gstrtpsession: Implement a number of feedback packet statistics 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711693
 2013-11-15 15:21:19 +01:00
Wim Taymans
c8db05d610 rtpsource: update receiver stats for sender 
An internal sender in a session is also a receiver of its own packets so update
the receiver stats. Other senders in the session will use this info to generate
correct RB blocks in their SR reports.
 2013-11-07 16:24:30 +01:00
Olivier Crête
b9ceafe5af rtpsession: Demux RTCP buffers from the RTP stream 
If there are RTCP buffers in the RTP stream, process them as
RTCP. This way, we want receive streams following RFC 5761

https://bugzilla.gnome.org/show_bug.cgi?id=687657
 2013-09-13 16:25:49 +02:00
Wim Taymans
28e5f90988 rtpbin: use PacketInfo for the sender 
Avoid mapping the packet multiple times when sending RTP.
 2013-09-13 14:34:28 +02:00
Wim Taymans
a02c9473d8 rtpbin: store more in the PacketInfo 
Store all info in the PacketInfo so that we can avoid mapping the packet
multiple times.
 2013-09-13 14:34:28 +02:00
Wim Taymans
e5c789abd6 session: store more in the PacketInfo structure 2013-09-13 14:34:28 +02:00
Wim Taymans
47662f9ca4 rtpbin: RTPArrivalStats -> RTPPacketInfo 
Rename a structure because we are also going to use this for the sender
bits.
 2013-09-13 14:34:28 +02:00
Wim Taymans
f1106cde66 session: only update next check time when reconsidering 
Don't update the next RTCP check time in all cases but only when we
reconsidered. This avoids delaying sending a full RTCP packet when we
are doing early feedback.
 2013-08-27 09:55:52 +02:00
Wim Taymans
47065db0b6 session: add more debug 2013-08-27 09:55:52 +02:00
Wim Taymans
4b7bcc2ec1 rtsession: fix locking 
We need to take the session lock when getting and manipulating the
source.
 2013-08-26 11:50:27 +02:00
Wim Taymans
3f46527f75 rtpsession: add some more debug 2013-08-26 11:50:13 +02:00
Wim Taymans
5fe18ee432 session: add some docs 2013-08-19 22:04:49 +02:00
Wim Taymans
482dacfb54 session: handle NACK feedback and generate events 
Handle and parse the feedback NACK packets and generate a Retransmission
event for each NACKed packet
 2013-08-19 22:04:49 +02:00
Wim Taymans
48174164eb session: add NACK feedback in RTCP 2013-08-06 15:50:19 +02:00
Wim Taymans
50638b8106 session: handle Retransmission event and schedule NACK 
Handle the retransmission event from downstream and use it to schedule a NACK
request.
 2013-08-06 15:50:19 +02:00
Wim Taymans
0bddbd682d session: pass data to remove func 
Pass the data to the remove function because we are going to deref it when there
is pli or fir.
 2013-08-06 15:50:19 +02:00
Wim Taymans
3c82de59f9 session: use common send_rtcp method 
Reuse the send_rtcp method that already asks for the current time when
requesting a keyframe.
 2013-08-05 15:02:59 +02:00
Wim Taymans
3c14c6021c session: Don't use ClockTimeDiff for unsigned delays 2013-08-05 15:02:59 +02:00
Tim-Philipp Müller
7469cd3a4c rtpmanager: use generic marshaller 2013-08-04 11:03:07 +01:00
Wim Taymans
9613e481ad session: add FIR and PLI like other RTCP packets 
Add the FIR and PLI packets like the other RTCP packet instead of from the
on-sending-rtcp default signal handler.
 2013-08-03 00:33:24 +02:00
Wim Taymans
02359f9219 session: don't make buffer writable prematurely 
There is no reason to make the SR buffer writable at this point. This is better
delayed until needed.
 2013-07-26 22:31:41 +02:00
Wim Taymans
0261199fc4 session: ignore RTCP for inactive sources 2013-07-26 22:31:23 +02:00
Wim Taymans
a4b4ca53c0 session: small cleanup 2013-07-26 22:25:17 +02:00
Wim Taymans
e0abd2e9b5 session: handle partial RTCP report blocks 
When we have more SSRCs to report than what fit in an RTCP packet, use a
generation counter to make sure all of them end up in a packet eventually.
 2013-07-26 17:29:10 +02:00
Wim Taymans
6cce6fb04c session: create SSRC before doing session cleanup 
Make the internal source before we do session cleanup
 2013-07-26 17:29:10 +02:00
Wim Taymans
5b0298c63e session: reorganize the report block code 2013-07-26 17:29:10 +02:00
Wim Taymans
3c44cd7c83 session: refactor active and sender checks 2013-07-26 14:21:40 +02:00
Wim Taymans
e952f7ba43 session: remove internal sources on timeout 
When an internal source times out and becomes a receiver, remove it.
 2013-07-26 12:18:01 +02:00
Wim Taymans
e9e2fe3950 session: create an internal source for RTCP 
When we need to do RTCP and we don't have an internal source yet,
make one.
 2013-07-26 12:18:01 +02:00
Wim Taymans
bd0709c15c session: remove old code to change SSRC 
Remove code used to change the SSRC after a collision. We now send
a RECONFIGURE event upstream to make the upstream element change the SSRC.
 2013-07-26 12:18:01 +02:00
Wim Taymans
abc90da1dc session: delay allocation of internal source 
Allocate the internal source when we receive a caps with the SSRC or when we see
a buffer with the SSRC.
 2013-07-26 12:18:01 +02:00
Wim Taymans
e0a1ce1291 session: generate reconfigure on collision 
When we detect a collision, change the SSRC that we suggest upstream
and trigger RECONFIGURE. This should make upstream select a new SSRC.
 2013-07-26 12:18:01 +02:00
Wim Taymans
495d43c089 session: produce RTCP for all internal sources 
Loop over all the internal sources and produce RTCP. We also need
to queue the RTCP packets and send them when we are finished.
 2013-07-26 12:18:00 +02:00
Wim Taymans
9505fd4150 session: deprecate internal source and ssrc properties 
Deprecate the internal source and internal ssrc properties. There might
be more than one internal source.
 2013-07-26 12:17:59 +02:00
Wim Taymans
3d6ee1fb5e session: internal sources don't use probation 2013-07-26 12:17:59 +02:00
Wim Taymans
0e53e9109e session: give caps to session 
Let the session parse the caps and update its SSRC when needed.
 2013-07-26 12:17:59 +02:00
Wim Taymans
c06482a2cb session: make method to suggest available SSRC 
Make a method to suggest the best available SSRC. This is the SSRC of the last
created internal source and is used to instruct upstream to produce this
SSRC.
 2013-07-26 12:17:59 +02:00
Wim Taymans
33ce50e8b1 session: keep SDES and set on new internal sources 
Keep track of the SDES ourselves and set it on all newly created
internal sources.
 2013-07-26 12:17:59 +02:00
Wim Taymans
5652f02b76 session: make method to make internal sources 
Add a method to obtain an internal source and use it to create
our internal source
 2013-07-26 12:17:59 +02:00
Wim Taymans
7f83927c95 session: count internal sources and how many are senders 2013-07-26 12:17:58 +02:00
Wim Taymans
719343c206 rtpsession: separate BYE marking and scheduling 
First mark sources with BYE and then schedule the BYE RTCP message.
 2013-07-26 12:17:58 +02:00
Wim Taymans
391943ba82 session: get SSRC from RTCP packet itself 
Get the SSRC from the RTCP packet instead.
 2013-07-26 12:17:57 +02:00
Wim Taymans
a3f75a17ef session: fix bandwidth calculation 
We iterate over all sources and the internal one is also in the
hashtable so avoid adding it twice.
 2013-07-26 12:17:57 +02:00
Wim Taymans
9eaef9d332 session: add some docs 2013-07-26 12:17:56 +02:00
Wim Taymans
2163355a47 session: Rearrange RTCP reporting a little 
Make a function to generate an RTCP packet for a source, pass the source as a
parameter.
Move timeout of collisions to session cleanup phase.
 2013-07-26 12:17:56 +02:00
Wim Taymans
a3bf374351 session: move check for is_early around 
Move the check for the early RTCP to where it is needed and used.
 2013-07-26 12:17:56 +02:00
Wim Taymans
b069db6a2e session: parse packet outside of the session lock 2013-07-26 12:17:56 +02:00
Wim Taymans
57c27ec319 session: do nicer checks for internal sources 2013-07-26 12:17:56 +02:00
Wim Taymans
93d07298ff session: let source keep track if it sent BYE 2013-07-26 12:17:56 +02:00
Wim Taymans
1d02496d15 source: also use the source for bye_reason 
Store the BYE reason in our internal source object. Rename the methods on the
source object a little because now the BYE can be received in RTCP or
set when the session wants to send BYE.
 2013-07-26 12:17:56 +02:00
Wim Taymans
ddd071e54c session: configure sdes with structure only 
Remove code to configure the SDES with methods and types, only
allow configuration with GstStructure
 2013-07-26 12:17:55 +02:00
Wim Taymans
0060e1d45d session: refactor add and find source 
Make functions to find and add a source to the hashtable.
 2013-07-26 12:17:55 +02:00
Wim Taymans
adb0d68c07 session: remove source from sync_rtcp 
We don't need to know the sender source of the session in the
callback, the SR packet is for all participants in the session.
 2013-07-26 12:17:55 +02:00
Wim Taymans
dece8413ef rtpsession: don't use invalid times in RTCP timeouts 
An invalid timeout can be calculated when we disabled RTCP by setting the
bandwidth to 0. Make sure all code can handle this case.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=674626
 2013-07-23 17:41:48 +02:00
Wim Taymans
25e0f0d6b6 rtpsession: lock session when changing bandwidth 
Take the session lock when changing the bandwidth properties so that we don't
end up with inconsistent behaviour.
 2013-07-23 17:41:48 +02:00
Wim Taymans
c337265ee4 session: reset some RTCP variables 
The early_send time was set to 0 and always triggering an early RTCP packet.
 2013-07-23 17:41:48 +02:00
Wim Taymans
51c9f7989f rtpsession: Use the right hashtable to calculate bandwidth 
Don't use an unused hashtable to iterate source to calculate bandwidth.
Remove unused code.
 2013-06-13 16:02:19 +02:00
Paul HENRYS
10802cae73 rtpsession: Fix wrong code organisation in case of collision 
change_ssrc field of RTPSession should be set before calling
rtp_session_schedule_bye_locked () as this function will call reconsider function
that will wake up rtcp_thread which will call rtp_session_on_timeout () that will
check change_ssrc to change the ssrc.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=694184
 2013-02-22 09:28:07 +02:00
Wim Taymans
2971ed44ee rtpsession: remove dead code 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=668355
 2013-02-07 15:06:40 +01:00
Wim Taymans
6b7d05ac57 rtpsession: improve debug 2013-01-07 14:22:48 +01:00
Tim-Philipp Müller
230cf41cc9 Fix FSF address 
https://bugzilla.gnome.org/show_bug.cgi?id=687520
 2012-11-04 00:07:18 +00:00
Wim Taymans
d5fd524a0c rtsession: fix compiler warning 2012-10-17 13:55:45 +02:00
Wim Taymans
f17db5c4ed rtpsession: update caps in the source 
Inform the source when caps changed. This was removed in the port to 1.0
leaving the source unaware of the clock-rate and unable to interpollate
rtp timestamps for SR packets.
 2012-10-17 13:22:40 +02:00
Wim Taymans
5b394385b9 session: also stop probatation on existing sources 
Receiving an RTCP packet should also stop probation on sources we have seen
before.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=683065
 2012-08-30 22:07:24 +02:00
Aleix Conchillo Flaque
4a200b670f rtp: make rtp packet probation configurable (bug #682512) 2012-08-30 21:49:57 +02:00
Wim Taymans
51371d26ee update for RTP buffer api changes 2012-07-17 16:38:27 +02:00
Pascal Buhler
8161daef4a rtpsession: creation should be signaled before validation 
https://bugzilla.gnome.org/show_bug.cgi?id=667850
 2012-05-09 10:36:18 +02:00
Wim Taymans
af59f573b5 rtpsession: don't leak the address 2012-03-13 19:26:47 +01:00
Mark Nauwelaerts
f189f62b13 Merge branch 'master' into 0.11 
Conflicts:
	ext/wavpack/gstwavpackenc.c
	tests/check/elements/audioiirfilter.c
	tests/examples/v4l2/probe.c
 2012-03-01 11:29:50 +01:00
Edward Hervey
9beda57c3a Suppress deprecation warnings in selected files, for g_value_array_* mostly 2012-02-27 14:47:25 +01:00
Wim Taymans
225e98d623 Merge branch 'master' into 0.11 
Conflicts:
	ext/flac/gstflacenc.c
	ext/jack/gstjackaudioclient.c
	ext/jack/gstjackaudiosink.c
	ext/jack/gstjackaudiosrc.c
	ext/pulse/plugin.c
	ext/shout2/gstshout2.c
	gst/matroska/matroska-mux.c
	gst/rtp/gstrtph264pay.c
 2012-02-10 16:23:14 +01:00
Olivier Crête
b850741430 rtpsession: Keep the buffer mapped while it is being modified 2012-01-27 19:05:23 +01:00
Olivier Crête
aeec2d5f7e rtpsession: Initialise the address pointer to NULL 2012-01-27 19:05:23 +01:00
Tim-Philipp Müller
5525e40970 rtpmanager: don't pretend our random hostnames are fully-qualified domain names 2012-01-25 13:19:12 +00:00
Sebastian Dröge
0b517ce9fb Merge branch '0.11' of ssh://git.freedesktop.org/git/gstreamer/gst-plugins-good into 0.11 2012-01-25 12:49:34 +01:00
Sebastian Dröge
10554b271f Merge branch 'master' into 0.11 
Conflicts:
	ext/flac/gstflacdec.c
	ext/jpeg/gstjpegenc.c
	ext/pulse/pulsesink.c
	sys/v4l2/gstv4l2src.c
 2012-01-25 12:49:11 +01:00
Wim Taymans
b4630dd3e0 more memory API porting 2012-01-25 12:30:29 +01:00
Tim-Philipp Müller
a476d529d2 rtpmanager: don't reveal the user's username, hostname or real name by default 
Send a randomly made-up user@hostname as CNAME and don't
send a NAME at all by default.

https://bugzilla.gnome.org/show_bug.cgi?id=668320
 2012-01-23 13:47:08 +00:00
Wim Taymans
1584806634 port to new gthread API 2012-01-19 11:33:53 +01:00
Sebastian Dröge
cb789e32ad rtpmanager: Port to GIO 2012-01-17 13:08:42 +01:00
Sebastian Dröge
93e3ed5a86 Merge branch 'master' into 0.11 
Conflicts:
	ext/cairo/gsttextoverlay.c
	ext/pulse/pulseaudiosink.c
	gst/audioparsers/gstaacparse.c
	gst/avi/gstavimux.c
	gst/flv/gstflvmux.c
	gst/interleave/interleave.c
	gst/isomp4/gstqtmux.c
	gst/matroska/matroska-demux.c
	gst/matroska/matroska-mux.c
	gst/matroska/matroska-mux.h
	gst/matroska/matroska-read-common.c
	gst/multifile/gstmultifilesink.c
	gst/multipart/multipartmux.c
	gst/shapewipe/gstshapewipe.c
	gst/smpte/gstsmpte.c
	gst/udp/gstmultiudpsink.c
	gst/videobox/gstvideobox.c
	gst/videocrop/gstaspectratiocrop.c
	gst/videomixer/videomixer.c
	gst/videomixer/videomixer2.c
	gst/wavparse/gstwavparse.c
	po/ja.po
	po/lv.po
	po/sr.po
	tests/check/Makefile.am
	tests/check/elements/qtmux.c
	tests/check/elements/rgvolume.c
 2012-01-10 14:32:32 +01:00